VideoClock works great, I guess, I do not see any immediate change in video performance.
Problem is - I am running ConvolverVST plugin, which runs a multichannel room correction filter. JRiver is the only media player that has let me do this in a stable manner and with ASIO. Now, this filter requries an audio stream with the same sample rate as the filter was recorded in. Without knowing any technical details aout VideoClock except that it is "slightly altering the audio sample rate", I would say that VideoClock in connection with a filter is a non-match. The result is that there are clearly audible artefacts that sounds like audio being out of phase. Typically, a speach from the centre channel sounds like it is moving right and left in the room every couple of seconds. Makes sense, really, if the sample rate of the signal to be filtered does not match the filter.
I am not used to 24fps, since my video card does not allow this. But this is about to change, I am curious about 24fps playback, since this rate is the only acceptable one to many video-philes.
Is it possible to alter the filter graph so that the re-sampling is performed after all other DSPs? Or will that do any difference at all?
Can anyone give a good explanation for what audio degradation to expect when resampling by a non-integer factor? To my understanding, resampling from 48kHz to e.g. 96kHz or 24kHz is straight forward, but from 48kHz to 46.789926kHz to fit the slowed down video may be more of a challenge. Anyone?